This invention generally relates to the digitization of continuously varying signals which are subsequently reconverted to continuously varying signals. More particularly, this invention relates to compression of digitized analog signals and subsequent expansion of the compressed digitized analog signals for reconversion to analog signals.
In the telecommunications field there has been a continuing trend toward the use of digital transmission techniques wherein analog signals, particularly voice signals, are digitized, transmitted and then reconverted to a sufficiently accurate representation of the original voice signal to be delivered at the receiving end of a telecommunications link. Such digitization of voice signals has permitted multiplexing a large number of voice band channels together, and the switching of those multiplexed voice band channels can then be performed economically using digital techniques. Presently, these voice digitization techniques have been applied primarily to long distance transmission wherein transmission costs dominate and significant savings are realized by utilizing digital transmission techniques of digitized voice signals. As the size and cost of microprocessors and memory keep decreasing it is becoming economically feasible to extend these digital techniques for processing analog signals into the terminal equipment at both ends of a telecommunications link. Within ten to fifteen years it is estimated that digital transmission techniques and equipment will completely dominate the telecommunications field from the subscriber terminal equipment through switching equipment to the long distance transmission equipment.
To further extend the usefulness of digital techniques in the telecommunications field much work is being done to reduce the number of bits in the digital signal representing a speech signal without impairing reproduction of the voice signal that the digital signal was originally derived from. This is done by signal processing of the continuously varying voice signals and the digital signals derived therefrom to achieve the bit-rate reductions. Different techniques have been developed for the bit-rate reduction of digitized voice signals which are generally referred to as waveform coding techniques. These waveform coding techniques include adaptive differential pulse code modulation, sub-band coding and transform coding, all of which can typically reduce the bit-rate of a digitized voice signal by more than a factor of two. Another technique used in the Bell System is time assignment speech interpolation (TASI) in which silence intervals within speech are detected and not transmitted. Yet another technique is vocoding in which speech is analyzed to extract its essential parameters followed by synthesis to reconstruct the speech. Presently, vocoder techniques do not offer the naturalness of speaker characteristics and its use is limited to applications where extremely low bit rates are dictated, such as in secure voice transmission.
Speech processing and digitization to reduce the number of binary bits making up the digitized voice signal are also being applied to a relatively new field wherein digitized voice signals are stored and then read out for reconstruction into voice signals at a later time. This is a store-and-forward approach wherein voice messages are digitally stored for later delivery to a receiving party via the telephone, similar to the service provided by electronic mail. Store-and-forward systems have been developed by a number of companies including Wang, IBM, Bell System and VMX, Inc.
In the store-and-forward systems large capacity memories are required and presently relatively expensive disk storage is utilized. Without bit-rate reduction techniques being applied to a digitized voice signal, speech may typically be digitized at 64 kilobits/second and every two minutes worth of digitized speech requires one megabyte of memory space for storage. It can readily be seen that extensive amounts of memory are required for a viable store-and-forward system and there is a need for techniques to reduce the amount of memory required to store a given amount of digitized voice. One approach has been to store only the active speech but this provides only a small amount of signal compression. The previously mentioned adaptive differential pulse code modulation, sub-band coding and transform coding techniques can compress the digitized signals by a factor of two which is a significant savings but there is still a need in the art to further compress digitized voice signals to minimize the amount of memory required to store these signals.